Google any random speaker frequency response graph. Look at the left to figure out where -4db starts and draw an imaginary horizontal line. If you move this line downward to -10 you will notice suddenly the speaker has a better low end response reaching further down to 0. Same goes for the high frequencies.

Now, sound like ripples in water doesn't just cut off suddenly or fade. Every ripple influences all nearby frequencies until infinity in the highs or absolute zero in the lows. So all speakers could in theory reproduce all frequencies down to near 0 and up to 20khz or more even your iphone in-ears cpuld (ehem) :D.
But as a normal human being you won't hear it because 10hz at -50db is just too low to notice.

BTW that last statement is false. IBecause if your speaker driver is big enough (8") and loud enough. Not your ears, but your furniture will notice.

Google any random speaker frequency response graph. Look at the left to figure out where -4db starts and draw an imaginary horizontal line. If you move this line downward to -10 you will notice suddenly the speaker has a better low end response reaching further down to 0. Same goes for the high frequencies.

No, I think you're reading it exactly the opposite of how it's supposed to be read to be honest.

Not at all.
MANY people know exactly what it means. Some people are away with the faeries but you do need to know the relevance of the two (and there are other) reference levels to the job you are doing.
Some gear produces it's 'best' performance at -10dBV signal level, and some at +4dBu.

+4dBu out of one piece of gear risks overloading (clipping) a -10dBV input.
-10dBV output of a piece of gear risks unnecessary noise (typically hiss) when going into a +4dBu input. If it is a unit that is calibrated for level, compressor , limiter etc, the threshold and other controls will not work at the scale points marked.

Not at all.
MANY people know exactly what it means. Some people are away with the faeries but you do need to know the relevance of the two (and there are other) reference levels to the job you are doing.
Some gear produces it's 'best' performance at -10dBV signal level, and some at +4dBu.

+4dBu out of one piece of gear risks overloading (clipping) a -10dBV input.
-10dBV output of a piece of gear risks unnecessary noise (typically hiss) when going into a +4dBu input. If it is a unit that is calibrated for level, compressor , limiter etc, the threshold and other controls will not work at the scale points marked.

So... the max volume before clipping in a -10dBV system is 1.78 Volts to around 3.16 Volts. But, the nominal Level in a +4dBu system is 1.23 Volts, so you can see... sending a +4dBu test tone into a -10dBV system is almost at the point of clipping already. Any spikes from drums or a loud vocal will easily distort the input of the -10dBV system because it is designed to operate on a much lower range/scale of voltage.

In comparison, the +4dBu system doesn't start clipping until somewhere between 6.9 Volts up to around 12.28 Volts. So you can see how a +4dBu system can handle A LOT more level. Why is that good? Because in Analog circuits that can help overcome RF and EMF interference that the copper wiring inside the circuit catches. The end result makes the +4dBu circuits/gear "sound" quieter since the audio signal passing through it is so much louder than the noise being inducted into the circuit from the air.

But the components needed to handle that amount of voltage effectively are always going to be more expensive than the components that aren't designed to handle that much voltage... this is where the "consumer vs professional" delineation came to be between -10dBV and +4dBu.

Theres nothing wrong with trying using resistors to match your impedances if you are the patient, careful and mindful type, but the common way is to use the gain knobs on a mixing console, or a direct box/preamp.

A potentiometer is a variable resistor, by definition, so u could just throw a 100 or so ohm pot on the cassette outs, in parallel to the audio signal, and adjust it close to wide open to get around that 50 ohms. You could just as well try a 8 or 6 ohm resistor in parallel to audio signal as a temporary fix. This essentially would be adding gain into the interface so you dont have to push either the tape main outs or the input volume sliders past unity (about 4/5 up).

Proper gain would adjust the input resistance- make the input more inviting and accommodating to the current. Using resistors in parallel on the tape audio out would draw more current out of the output stage of the tape, and on the factor of 600 ohms to 50 ohms, it could cause the audio output circuit to overheat- thats doubtful to damage any components but would introduce distortion , so, if u go that way, use potentiometers so you can bring it right to barely distorting, then bring it back clean.

You just dont want any of your volume sliders wide open- thats asking for background noise.

A preamp/direct box with ground lift would be the best for avoiding em interference type background hiss, but its not necessary in every building. In a home setting with no bad ground, a stable power grid, and not nextdoor to a radio broadcast tower- you just have to keep your input cables from running parallel to any power cables. The natural noise floor of 24 bit recording isnt gonna be audibly noticeable, even at -10, but if you crank all the ins and outs till 11 you will introduce noise; hence a mixer or preamp with gain, to match impedances. Personally, id use a small, high quality mixer in front of your interface- but thats only because my real preamp / direct box seems to have died last week... A pair of potentiometers and a small project box sounds fun, as well. Just make sure you account for the total amperage, wattage- not just ohmage. Its doubtful youd burn out the pot just attenuating a -10 db line level audio signal; however, if you exceed the wattage rating of a pot or resistor, youre askin to start a fire. If the output stage of your tape player uses all 2(+) watt resistors, and decent capacitors, you should be fine overdriving it a little with some parallel resistors or a potentiometer- its those mosfet ic''s that burn out first, and your tape player shouldn't have any of those

. You are looking at approx 600 ohms,-10db,for a microphone or guitar input, and around 50 ohms,+4db, for professional electronic instruments. And we havent touched trs yet. The actual outboard recording and top line PA hardware uses trs , three conductors, 1/4 inch, like an xlr mic cable, that cancels out em interference and noise, and rates at +12 db continuous- thats four times louder than the +4 ts 1/4", at the same noise floor. TRS is only used for serious pro hardware recording and the top PA gear because the sheer amount and lengths of cables in these settings need special precautions against picking up em interference.

None of my instruments- synths, samplers, drum machines, guitars) have trs ( looks like a stereo 1/4 jack), because the unbalanced guitar cables ( mono 1/4") and RCA pairs arent near long enough to run into so much em interference.

My bass amp has all unbalanced ins, but it has a balanced record out- because that record out could possibly have to run 100 yards back to the recording setup, alongside 30 other audio signals, all running back to the sound booth together.

Idc about dbv, dbw whatever. Decibels are defined on subjective opinions. One decibel is hardly noticeable, 2 decibels are very noticeable, and an increase of three decibels is perceived as a doubling of sound pressure/ volume. If u could reach +4 db nominal, your inherent noise floor would drop fourhold- but even at 4x noise floor at -10 is not going to be noticed in 24 bit, unless you are cranking your tracks to peak at 140 db in an arena hall.

I trust you realise that most of your post is complete twaddle.
There is so much in there it is hard to know where to start however { Idc about dbv, dbw whatever. Decibels are defined on subjective opinions.} tells everyone else what they need to know.

Sp 1200 should be +4, but thats all dependant on getting your samples recorded at the perfect level. And with my experience with vintage samplers, a single sample always plays back too quiet because the sampler reserves headroom for playing back multiple samples simultaneously. I rumln my mpc's thru an analog mixer with gains, so the +4-10 is a non issue. The 1200 is +4, but when using only one or two samples at a time , it would likely fall more into the -10 range.

Consumer soundcards dont have -10 or +4 main stereo outs- that jack is pushed by a 3 watt headphone amp, and many soundcard makers boost the 100-120 hz range, add muddy verb/delay and compress the audio by default in the soundcard advanced options utility.

You will be fine using it as is at +4, but you might want to check advanced sound settings to see if your sound is colored and boosted, disable any of that, turn the volume down around 5, and try -10 again.

Quote:

Originally Posted by chazfilez

After reading through this very informative thread, I am stumped as tot he output impedance of my my E-Mu SP-1200. The manual sheds no light on whether 1/4" unbalanced outputs are +4/-10. Do any of you OGs have thoughts or knowledge of what an old 12-bit sampler of this age would have been set-up for? I am running into a RME Fireface UCX which has the option for eith +4 which is obviously lower level and -10 which sounds "hotter." Other than just doing what "sounds good" does anyone have insight for this specific scenario going direct from sampler to audio interface? Note: I go out of the eight channel outputs with a Y_cable (TRS-TS/TS) to take advantage of internal filters on the ring (return). Thanks in advance!

Quote:

Originally Posted by orior

Hello everyone,

I've just read this thread and my head hurts trying to understand all this technical stuff.
I just need to know the right setting for my case, so I don't blow up my speakers, and I got really confused trying to figure the correct ones.

I've just bought a pair of jbl lsr305 monitors.
The default setting is -10db.
I've connected it using this cable:Musician's Friend
(Only in my cable the red ones says "Ring" and the black ones says "Tip")
It's connected to the output of Creative soundblaster Z.

At -10db in the speakers The sound is really loud, to the point it even sounds compressed. even old low volume songs sound very punchy.
I've switched to +4db, and now it's not as loud. It's less punchy but sounds quite right.

Depends on the physical construction of the woofer and box and the quality of the signal being played.

A properly designed folded horn cabinet with a high quality, high spl ( high efficiency per watt) subwoofer can exceed 140 db(loud as gunshot, point blank range) at 200 watts rms/ 400 peak without blowing, given the speaker is properly designed physically, the cabinet is the proper size for that particular woofer, and the amplifier is producing an undistorted signal.

200/400 watts is relatively nothing. Ive seen a subwoofer that had NO rms, max, program or peak rating, because the designer hooked the prototype up to a 20,000 watt monoblock that he had designed to accompany it, and he couldnt manage to either blow the speaker, or peak out the amplifier.

Db, sound pressure, is all dependent on the area that is being filled with sound. Ive never heard a single 20,000 watt sub, but ive been in a 12,000 seat arena where the system had about 20,000 watts going through about 16 subs, and, as huge as that arena was, when the bass dropped, it was at least 135 db in the bass, way out in nosebleed. Db drops by 6 db, or fourfold, every time you double the room size, i believe. Now if that arena was easily 50 times larger than a movie theatre, those 20,000 watts that were hitting 130 in the arena, would exceed 400 db in a relatively smaller movie theatre.

. 20,000 watts would be that super unblowable, ooak woofer.That would likely be lethal to anyone present, as a mere 160 causes complete hearing loss and intense physical pain already. At 400db, even if it didnt squish your innards, or stop your heart, it would make breathing impossible, and anyone would suffocate- at the least.

Please correct me on the formula relating sound pressure with volume... But thats an example of a one of a kind, godzillawoofer. 20,000 watts, and possibly over 400 db- the thing is totally unblowable, and it also has @ a $20,000 price tag. The matched 20,000 watt amp was $30,000- but it did come with a second, identical free amp. Guess somebodys gotta repay their r&d budget...

The speaker, physically, needs a frame that wont flex, sufficient room for excursion ( so the former { tube that the voice coil is wired around} doesnt recoil from a hard kick, and bottom out- causing the former to bend and make a scratching sound when the speaker basses) a bent former is one kind of blown, caused by too much volume/displacement. The db point for this widely varies between speaker models and designs.

If the former has sufficient room for displacement, and gets that huge bass hit that made the other speaker bottom out, it could do the opposite and tear the spider ( flexible, round fabric that holds the former and its voice coil dead center of the speaker magnet), and, if it tears thru the spider, it will likely tear thru the center of the speaker cone, maybe even right thru the dust cover, and poke out thru the cone of the speaker. If the former ( which holds the voice coil) is ejected from its home within the speaker magnet, the audio signal running thru the voice coil will produce no sound. If the coil isnt centered in the magnet, the speaker will not vibrate or create sound.

The former, voice coil, audio current and speaker magnet all work together to make an electromagnet which can vibrate an attached diaphragm and turn electrical current into audio. With a cheap sub, you can also tear the surround if you exceed displacement. If the speaker is physically solid, but you grossly overpower, or feed distorted audio to it, the voice coil can melt. Current creates heat dependent on its medium, but 1% distortion can blow a speaker at a 1/4 of its rated power.

I had a fosgate series 1 200 watt 15" former push right thru the dustcap on me, when i put it on a punch 60 with 600 watts peak power. It was rated 0.03% thd at peak. Distortion didnt kill it, nor did current. It was simply unable to physically move that far in and out without tearing itself apart. That would have been around 125 dB, but that was a series 1 on a punch series amp. It was just a subpar driver. But for 20,000, you can buy a sub that will blow your intestines far before you ever blow that subwoofer.

And how to convert a +4 output in a -10 signal? I want to connect my mixer balanced monitor out to my speakers amp (rca). My mixer out converts the output in unbalanced if you plug a TS but i think the signal level is the same, +4db.

You could get a di box, some dual compressor limiters can link tr, xlr, +4 and -10 like my behrenger ultragain pro.

You could do the math- something like an 800 ohm dummy resistor pair, or a pair of 1000 ohm potentiometers, bit theres another formula to see how many watts the pots or resistors need to off balance.

You could just always keep the monitor volume send below 1/4.

Forget about all the dB dBv dBu, the short of it is, in the golden radio days, microphones were all unpowered, and around 600 ohms- highly resistant to current, so they were quieter. Instruments had less electrical resistance, 50-100 ohms for an electric guitar or theremin, and were thus louder. The +4 dBv and -10 dBu were ideals- hardly precise standards. So now, every decent mixer has a wide sweeping gain knob, every ( most every) direct box/ preamp has a gain, and daws have normalize features.

A pc and a focusrite solo makes a good drawing board- great for doodling and trying out new ideas, but computers and 1-2 channel interfaces are cheap, anybody can get them- which is great, but they hardly make a full working studio unless you are an unbelievable solo guitarist and dont play nothin else, or u just make really simple music

I think thats to others. U got a mixer- hardware, good. I seriously wouldnt worry about converting +4 to -10, when.you can just pull the monitor sends down to 1/4. But, since u like hardware- do check out the behringer ultragain pro. Ts unbalanced 1/4and xlr balanced ins / outs, and -10/+4 switch for both channels, and limiters!!!

i am not even going to attempt to read whole thread... did any one actually provide PRACTICAL information for planet earth and reality?... cuz it all seemed like everyone was quick to melt this guy's mind with super deep logarithmic equations and fancy-book learnin... when honestly there was prolly a quick PRACTICAL, USEABLE small set of info that would have had this guy off and running in minutes.

EDIT: as far as LOOONNNNNNG posts... yeah, you have plenty of intelligent people that are trying to make music... I think what happens is that they are not exactly successful yet in their musical ventures, and so come here and bloviate sweet information; moreover, causing them to walk away with a stroked ego... an ego, stroked by themselves.

EDIT2: my thought is that some of these guys are amazing electrical engineers, but utterly suck at making music... otherwise theyd rather be working on a track then rewriting encyclopedias.

i am not even going to attempt to read whole thread... did any one actually provide PRACTICAL information for planet earth and reality?... cuz it all seemed like everyone was quick to melt this guy's mind with super deep logarithmic equations and fancy-book learnin... when honestly there was prolly a quick PRACTICAL, USEABLE small set of info that would have had this guy off and running in minutes.

EDIT: as far as LOOONNNNNNG posts... yeah, you have plenty of intelligent people that are trying to make music... I think what happens is that they are not exactly successful yet in their musical ventures, and so come here and bloviate sweet information; moreover, causing them to walk away with a stroked ego... an ego, stroked by themselves.

EDIT2: my thought is that some of these guys are amazing electrical engineers, but utterly suck at making music... otherwise theyd rather be working on a track then rewriting encyclopedias.

Why do home burned CDs sound lower than professionally burned ones.? I set my tracks in garage band to peak at 0db. Yet when I export to mp3 or aac and burn an audio cd in iTunes its quieter than I expect. Is it due to mac garageband being not a pro system. Ie is what I think is Odb actually being saved as _10db and not +4db? If you get my meaning..

Why do home burned CDs sound lower than professionally burned ones.? I set my tracks in garage band to peak at 0db. Yet when I export to mp3 or aac and burn an audio cd in iTunes its quieter than I expect. Is it due to mac garageband being not a pro system. Ie is what I think is Odb actually being saved as _10db and not +4db? If you get my meaning..

"+4" and "-10" that's discussed in this thread doesn't apply to digital at all.

Most likely what you hear is a difference in perceived loudness which is different from sample peak values. So if you set your absolute sample peak to 0dBFS in Garage Band that doesn't really tell you how "loud" it is perceived.

During mastering music will be adjusted and often made to be perceived as more loud. So if you don't do that then that's why it's different, most likely.

Because professionally produced material is usually compressed or otherwise manipulated to make it 'sound' as loud as possible.

A digital system has no headroom, of itself. Full scale digits is as 'loud' as it can ever get.
The analogue circuitry that precedes or follows A/D or D/A conversion will have some headroom.
Incidentally one of the definitions of analogue headroom specifies the allowable distortion, thus in early electronic gear 'clipping' was defined at say 1% or 0.1% total harmonic distortion.
In the 1950's, HiFi was anything lower than about 5% distortion.
The power rating used by many guitar and other musical instrument amplifiers can still use up to 10% distortion as a reference 'degradation' when specifying it's maximum output level.

There are various laws of physics that have relevance to level changes between a nominal -10dBV system to a +4dBu system as amplifying 'quietly' (adding no extra unnecessary noise) or even attenuating, is slightly problematic although most of the time the benefits outweigh the small disadvantage.

If I remember correctly, the earliest HiFi CD players would often produce a 2 Volts RMS output when the audio on the CD was at the maximum DIGITAL level possible ( which is all bits at '1').
Matt S

Possibly!
Your logical thinking is correct BUT you need to check the specifications of your convertors as interface/convertors that will actually handle +24dbu are fairly rare, with many being 'only' +22dbu or even +20dBu maximum before their inputs will clip.
Thus if you use usual analogue VU meters you may only have 16dB of headroom before the convertor clips.
Since most analogue mixing desks and other peripheral gear can output at least +21dbu unbalanced or +28dBu balanced you can start to appreciate where problems might creep in.
Of course 16dB headroom is often fine as music peaks are I believe typically around 12dB above an 'average' level depending on the nature of the signal of course, but the issue being that if your traditional wiggly VU meter is showing signal above the red zone, you have no idea what the peaks really might be although with practice you may learn by the speed that the needle disappears off the end of the scale!
Matt S